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Questions tagged [lowpass-filter]

A low-pass filter is an electronic filter that passes low-frequency signals but attenuates signals with frequencies higher than the cutoff frequency.

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1answer
832 views

How can I determine the order of filter?

This is my first study about signal analysing. I'm very confused about filter order. My problem is how can I know whether its 12-th order, or 2nd order like the book says so? I already knew that slope ...
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2answers
165 views

High / Low Pass Filters from Zero

I everyone, I'm a computer scientist but not from the signal processing community. I want to implement a real time digital low pass filter (to begin with) for a sound processing application. I've ...
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1answer
151 views

Help generating coefficients for basic FIR LPF

I'm trying to generate coefficients for a FIR low pass filter to be used in an FPGA. I'm using python (scipy.signal) to attempt to do this, but am having trouble getting coefficients in a usable form....
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399 views

how can I make a simple lowpass filter with this constraints? [closed]

I'm searching for a simple low pass filter code in C with this constraints a modulable frequency cutoff (how can I convert all these crazy coefficients into a frequency?) I could change with a slider....
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2answers
2k views

Gaussian filter as a low pass filter

Can anyone explain me how is a Gaussian filter a low pass filter? It may be a simple thing but I just can't seem to wrap my head around it. Also while applying a low pass filter for bandlimiting (to ...
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1answer
732 views

Consider an ideal low pass filter $H(\omega)$, and the input to this filter is the periodic square wave $x(t)$. Find the output $y(t)$

The solution to the problem is $$y(t) = 5 + \frac{20}{\pi} \sin(\pi t) + \frac{20}{3\pi} \sin(3 \pi t) $$ and to get that the solution says to find the Fourier series expansion of $x(t)$ and I am ...
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3answers
975 views

How to Design Nonlinear Phase FIR Filter?

I've recently encountered a rather clever low-pass filter that appears to have been designed to sacrifice uniform group delay in the pass-band to get better frequency response relative to the filter ...
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1answer
986 views

Filtering an acceleration signal

I have a triaxial accelerometer giving me acceleration along x, y, and z. I then combine the 3 axes by calculating magnitude of acceleration over all time points (details in my question here) I'm ...
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1answer
7k views

Differences between low-pass, band-pass, notch filters

I'm studying image processing and I know that there are different filters you can apply. I don't understand the difference between the low-pass/bandpass/notch filter (or high-pass/band-reject/notch). ...
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2answers
1k views

Resampling an audio signal

I am trying to resample an audio signal in my application. It will be between 48kHz and 96kHz audio (both upsampling and downsampling). Looking at the Wikipedia page for resampling I see there are ...
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1answer
491 views

Sampling the impulse response

Given two IIR filters (one Low-Pass and one High-Pass) designed using Butterworth, how can I know if they can be obtained using sampling the impulse response?
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1answer
6k views

How can I construct a Band-pass filter from a low and a high-pass filter?

Suppose, I need to construct a Band-pass filter in OpenCV. But, I know, there are no functions in OpenCV for Band-pass filters. Now, what I need to do is to have a low-pass filter and a high-pass ...
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3k views

Is there a set formula for calculating FIR coefficients (for a LPF ir 96 taps)?

I need to calculate a series of coefficients for several FIR filters having 96 taps is there a standard or set formula for calculating these, it is a low pass filter.
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204 views

Low Pass Filter - pixel correlation

I am trying to understand low pass filters for image processing, I came across an article that stated that low pass filters effect noise more than the real data due to that the image Noise always ...
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1answer
76 views

Why is the output of the LPF half that of the real/imaginary part of the complex envelope?

The passband signal $\ u_p(t) = I_{[−1,1]}(t) \cos 400πt $ is passed through an LPF as $\ u_p(t) \cos(401\pi t).$ The question is to find the output. The example does the following: If we let $\ u_1 =...
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245 views

Transform a Continuous Time Differential Equation of a Low Pass Filter to Discrete Time

Assuming I have a linear differential equation of first order low pass filter: $$RC*y(t)'+y(t) = x(t) $$ And Assuming i know now that the input x(t) is being sampled at a rate of $T_s$ how can now ...
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42 views

Denoising approach for a combination of several ADC voltage channles

I have 2 ADC channels of constant voltage measurement with small amount of high frequency noise only and no low frequency oscillations. The final signal should be the simple sum of those and denoised ...
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1answer
4k views

Why are ideal filters not particularly useful/ hard to do?

Why is a LPF ideal filter: a) Not useful? b) Hard to make? I was asked to make a Low pass filter (simulink block) that will take an input of a cutoff frequency, and for a signal with a frequency ...
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1answer
2k views

low pass filter with minimal ringing

I am trying to filter out some square wave signal to within a limited band (1/4 or 1/8 of the original), I realized that there's a lot of ringing in the wave when I use my filter (elliptical), I also ...
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2answers
6k views

Selecting the Low Pass FIR Filter, Decimation, DSP

I'm studying DSP, but I have weakness about it. I was reading this: http://www.eetimes.com/document.asp?doc_id=1275556 But, I don't know how select the Low Pass Filter. I am assuming that this ...
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2answers
2k views

Calculating Savitsky-Golay filter coefficients - normalization

Using the matrix equations found on Wikipedia I wrote some Python code to compute the Savitsky-Golay coefficients for $m$ data-points, using a polynomial of degree $k$. Here it is: ...
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1answer
304 views

How to filter uniform noise in a microcontroller?

By uniform noise I mean a band of a fixed width, no spikes, etc. This has to be one of the simplest examples. I'm looking for the appropriate theoretical approach as much as something relatively ...
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1answer
440 views

What is a low-pass filter for a sequence of tuples?

I am currently reading SEGMENTATION AND RECOGNITION OF SYMBOLS WITHIN HANDWRITTEN MATHEMATICAL EXPRESSIONS. There is one sentence I don't understand: The preprocessing stage is subdivided into ...
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1answer
116 views

Multiply FS Coefficients by 0 or 1 to get Low Pass Filter?

I have written a bit of code to upsample and interpolate a sample waveform in MATLAB. It's at the point where I have taken a signal, upsampled it 2x and filled in the gaps with 0's, and then I found ...
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1answer
263 views

How to choose one among different types of LPF?

My signal graph is as follows I have < 10 reputation, thus cannot post the photo... The graph are simply roughly the series of -5, 0, +5, 0, -5, 0, +5, 0.... but with sawtooth-like small peaks ...
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1answer
183 views

Approximation using a Fourier transform with low pass filter

I need to approximate a function f, but I cannot do so with frequencies that exceed 1kHz What is the best approximation I can get? Is taking the Fourier transform then zeroing any term above 1kHz the ...
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1answer
39 views

signed 2's complement of a negative number for Integer = 3 and Fraction = 9

I am working on signed 2's complement for my fractions coefficients. Below code can be used for positive inputs. However for a negative 2's complement when I added LSB with 1, then an error pops. I ...
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1answer
63 views

Moving Average Filter's Magnitude Response

I am working on filtering of data using the Moving average filter in MATLAB. My data is recorded temperature values on the Y-axis and Time(in seconds) on the X-axis (I'm getting the magnitude response ...
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1answer
35 views

A communication system

I am solving the following problem as an exercise. I have a signal $$z(t) = x(t)\cos(wt) + y(t)\cos(ut)$$ and we assume $x,y$ to be band-limited continuous time signals. In the frequency domain we ...
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2answers
77 views

Decimation Question about using a low pass filter before downsampling

I have a bunch of audio files that have a sample rate of 380kHz, however I would like to reduce this sampling rate to 44.1kHz. From doing some research online, it seems like I will have to use a low ...
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1answer
60 views

why does my integrator perform noise shaping?

I am designing (in matlab) a simple system with a DC signal at 0 Hz and some added noise. I am performing a differentiation of the signal (by simply subtracting the last values from each other), next ...
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1answer
44 views

How to derive low pass filter $\frac{N}{N+S}$

I watched the video for PID control system where it mentioned the Laplace domain function and low pass filter to be $\frac{N}{N+S}$. I used asymptotic analysis to see that it made sense. However, I'm ...
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57 views

Is a “Spectral Tilt Filter” the best way to get an arbitrary slope and cutoff LPF?

I need an arbitrary slope audio LPF for which both the cutoff frequency and slope in dB/oct can be smoothly modulated in real time. Is the "Spectral Tilt Filter" described by Julius Smith the correct ...
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1answer
81 views

y(t) a of an integrator circuit

I have a signal $$ x(t)= \frac{1}{T} e^{-\frac{t}{T}} u(t) - \frac{1}{T} e^{\frac{t}{T}} u(-t) $$ and I know that it transits in a integrator circuit and I have to find y(t) in time and frequency ...
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1answer
115 views

Low pass filter output showing instability

I am fairly new to DSP, and I'm trying to implement the Pan Tompkins algorithm for QRS detection of ECG signals in MATLAB. The first stage of the algorithm consists of a second-order low-pass filter. ...
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3answers
238 views

Reverse engineering a digital filter

I have 2 simultaneous signals that both are designed to measure eye movements. They are sampled at 250 Hz. We have 12 subject recordings. For 3 stable periods in each subject, we choose 256 points and ...
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1answer
71 views

how can I design Chebyshev filter with 5th-order polynomial and 0.001%-allowed ripples?

I'm very beginner in signal processing. I need to design a Chebyshev filter with 5th-order polynomial and 0.001%-allowed ripples by using matlab as I found in a paper. I'm filtering pressure data ...
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1answer
444 views

digital IIR filter delay and oversampling

Can oversampling decrease the delay of digital IIR filter? Imagine there is some digital signal going into processor that applies low pass filter.Lets say its 1 KHz sample rate and the filter is ...
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1answer
443 views

How to properly filter a BPSK demodulated signal?

I am new to this field and yesterday I began a project for BPSK modulation. So this is what i did: $\ x(t) = 0 $ or$\ 1 $ binary data $\ Fs = 15 $ sampling frequency kHz $\ Fc = 10 $ carrier ...
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1answer
122 views

How to filter a signal sampled at different frequencies

I have a signal which was sampled using different frequencies. The 3 regions are: 0-10 ms: 10000 Hz 10-50 ms: 1000 Hz 50-200 ms: 100 Hz The signal is very noisy, so I thought about smoothing it with ...
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1answer
88 views

Digital Lowpass Filters, basic questions

Forgive the newbie questions, I'm just starting out and wanted some basic information. I have a signal coming from a digital gyro, most of the useful information is below 90hz but I am seeing ...
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2answers
385 views

Determine filter type (HPF or LPF) by its coefficients

I want to determine optimal initial states for a FIR/IIR filter to get rid of oscillations and shifts at the initial interval. I've found that these initial states depend on filter type in the ...
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1answer
367 views

FIR Filters - Type 3

So I understand that a type 3 filter is not suitable for a highpass filter design, but is there any reason why it isnt suitable for a lowpass filter? So ultimately, can a type 3 linear-phase FIR ...
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1answer
748 views

RMS of absolute-valued sinusoid

A signal $$x(t)=\left|\sin(2\pi(1000)t)\right|$$ ($\left|x\right|$ is the absolute value function) is fed to an ideal low-pass filter with cutoff frequency of $1500\,\mathrm{Hz}$ to produce an output $...
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1answer
2k views

Interpolation filter gain

If somebody could clarify why everywhere is written that the interpolation filter should have gain $L$ where $L$ is upsampling factor. What this gain actually means? Does it compensates for the ...
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1answer
85 views

Short lowpass filter that also extrapolates lines

I'm looking for a short lowpass filter that also extrapolates lines, [0 1 2 3] --> 4, so I need (correct me) $H([\ 0\ \ 0.25\ \ 0.5\ ]) \approx [ 1\ \ 2\ \ 0 ]$ . How close can one get to this with 2 ...
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2answers
121 views

Transform one-pole cascaded system into a biquadratic section

I have two cascaded one-pole sections with the same coefficient b for both sections. $$ y_A[n]=x_A[n]+b\cdot(y_A[n−1]−x_A[n])\\ y_B[n]=y_A[n]+b\cdot(y_B[n−1]−y_A[n]) $$ The final output of the system ...
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1answer
334 views

Transfer function, amplitude response and difference equation for a filter

I've found a paper with a filter described in terms of transfer function, amplitude response and difference equation: transfer function of the second-order low-pass filter: $$ H(z) = \frac{(1-z^{-6})...
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2answers
3k views

Find Stopband Edge Frequency

I tried to find the stopband edge frequency.I know that the signal amplitude falls to -3 dB below the passband edge frequency.I examine a lot of DSP books so i don't any information about how to ...
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1answer
1k views

What is the low-pass filter that gives minimal transient response?

As title says, what is the low-pass filter that gives minimal transient response? By minimal transient response, I mean minimal settling time with reference to very tiny amount of divergence of steady-...

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